Use backticks not vertical bars to denote variables in comments for /modules/audio_device
Bug: webrtc:12338 Change-Id: I27ad3a5fe6e765379e4e4f42783558c5522bab38 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227091 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34620}
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WebRTC LUCI CQ
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@ -68,7 +68,7 @@ static const int kFilePlayTimeInSec = 5;
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static const size_t kBitsPerSample = 16;
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static const size_t kBytesPerSample = kBitsPerSample / 8;
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// Run the full-duplex test during this time (unit is in seconds).
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// Note that first |kNumIgnoreFirstCallbacks| are ignored.
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// Note that first `kNumIgnoreFirstCallbacks` are ignored.
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static const int kFullDuplexTimeInSec = 5;
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// Wait for the callback sequence to stabilize by ignoring this amount of the
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// initial callbacks (avoids initial FIFO access).
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@ -127,7 +127,7 @@ class FileAudioStream : public AudioStreamInterface {
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void Write(const void* source, size_t num_frames) override {}
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// Read samples from file stored in memory (at construction) and copy
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// |num_frames| (<=> 10ms) to the |destination| byte buffer.
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// `num_frames` (<=> 10ms) to the `destination` byte buffer.
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void Read(void* destination, size_t num_frames) override {
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memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
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num_frames * sizeof(int16_t));
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@ -171,7 +171,7 @@ class FifoAudioStream : public AudioStreamInterface {
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~FifoAudioStream() { Flush(); }
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// Allocate new memory, copy |num_frames| samples from |source| into memory
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// Allocate new memory, copy `num_frames` samples from `source` into memory
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// and add pointer to the memory location to end of the list.
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// Increases the size of the FIFO by one element.
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void Write(const void* source, size_t num_frames) override {
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@ -192,8 +192,8 @@ class FifoAudioStream : public AudioStreamInterface {
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total_written_elements_ += size;
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}
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// Read pointer to data buffer from front of list, copy |num_frames| of stored
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// data into |destination| and delete the utilized memory allocation.
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// Read pointer to data buffer from front of list, copy `num_frames` of stored
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// data into `destination` and delete the utilized memory allocation.
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// Decreases the size of the FIFO by one element.
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void Read(void* destination, size_t num_frames) override {
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ASSERT_EQ(num_frames, frames_per_buffer_);
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@ -251,7 +251,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
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rec_count_(0),
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pulse_time_(0) {}
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// Insert periodic impulses in first two samples of |destination|.
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// Insert periodic impulses in first two samples of `destination`.
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void Read(void* destination, size_t num_frames) override {
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ASSERT_EQ(num_frames, frames_per_buffer_);
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if (play_count_ == 0) {
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@ -272,14 +272,14 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
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}
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}
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// Detect received impulses in |source|, derive time between transmission and
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// Detect received impulses in `source`, derive time between transmission and
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// detection and add the calculated delay to list of latencies.
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void Write(const void* source, size_t num_frames) override {
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ASSERT_EQ(num_frames, frames_per_buffer_);
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rec_count_++;
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if (pulse_time_ == 0) {
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// Avoid detection of new impulse response until a new impulse has
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// been transmitted (sets |pulse_time_| to value larger than zero).
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// been transmitted (sets `pulse_time_` to value larger than zero).
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return;
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}
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const int16_t* ptr16 = static_cast<const int16_t*>(source);
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@ -298,7 +298,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
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// Total latency is the difference between transmit time and detection
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// tome plus the extra delay within the buffer in which we detected the
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// received impulse. It is transmitted at sample 0 but can be received
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// at sample N where N > 0. The term |extra_delay| accounts for N and it
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// at sample N where N > 0. The term `extra_delay` accounts for N and it
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// is a value between 0 and 10ms.
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latencies_.push_back(now_time - pulse_time_ + extra_delay);
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pulse_time_ = 0;
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